with this setup : caller->openser->asterisk->a2billing-> callee. I noticed the voice payload needs to go from the phone to a2billing, and then from a2billing to callee.

the bandwith usage is really a big issue here. for example, g711 codec will use 64k bps. since the same voice pay load needs to go from caller to a2billing, and then emerge from a2billing box to callee, essentially each phone call is costing 64K+64K bandwidth. even for a 1 MEG full duplex box, it can only support 1M/64K= 15 phone calls. with bandwidth so expensive in the colocation, it really cost lots of money to scale.

however, if we can get re-invite working, after caller knows callee, they talk directly to each other, then there is no voice payload going through a2billing. and the whole system can scale up drastically. is this possible?
if so, how can we get this done ?

and use round robin to switch the ip addresses or even making a little script that computes the load averages for all the nodes and gives the least loaded server ip to the user so under cloud hosting you can just set up your own template and grow as much as you want. Is it too simple? Am i missing a key point?

Well besides of doing sip2sip calls, then I would probably use 1 server for voip to voip calls only. Please tell me what you think..

OpenSER or SER has a number of advantages in terms of SIP registration over Asterisk, these are: -

1. For a given peice of hardware, SER can support many more registrations than Asterisk, but Asterisk can manage quite a few anyway, so not as bigger advantage as you might hope.

2. SER can load balance by sending say 20% of calls to one asterisk server and the other 80% somewhere else.

3. SER when combined with say MediaProxy sorts out all sorts of far end NAT problems, which are often experienced with just Asterisk. This is probably the main reason why it would be desirable to come up with a working config to combine A2Billing and SER, so that end users can use SIP devices behind all sorts of NAT without having to reconfigure firewalls and routers.

One way forward is to create a SER configuration that uses the SIP buddies table in A2Billing to authenticate its users. So SIP endpoints are registered to SER, not Asterisk, but in terms of adding new users and general management of customers, this should all be done in A2B.

When a SER registered SIP user wishes to make a call, the call is passed through to Asterisk, and the user recognised by A2Billing on the basis of their SIP address. (caller ID maybe?)

A2Billing then handles the routing and charging of the call.

Given that the A2Billing Database can support many Asterisk servers, and SER can deliver calls to different Asterisk servers, this would make for a highly scalable system.

So if anyone has intimate knowledge of SER, unfortunately I do not, then your challenge, should you chose to accept it, is to: -

1. Install OpenSER2. Combine MediaProxy module into the installation3. Create a config that uses the A2Billing SIP buddies MySQL table for SER authentication.4. Get it to pass through calls to Asterisk so that they are securely and automatically identified and authenticated by A2Billing.

and finally, tell us how you did it with sample configs and plenty of documentation, or get as far as you can, tell us where you get stuck and maybe with a bit of combined knowledge, we can make this config work.

Any offers? I'm sure many would be grateful, including me.

Yours

Joe

I have this setup only im using opensips.. and i couldn't get the media server too work.. and didnt configuire open sip...

Yes it can, create another sip peer and set the account code directive to the A2Billing account number.

Joe

Ohh yeah your right,. lol but still with this setup my clients can park calls while other clients can pic them up (not good) and they can dial each other by extension.... also i cant dish out the same extension multiple times

A2b stays in the media path of the call. meaning that a2b is not passive it is not called after the call is completed and say please compute the price. It is very active in the call. It calls asterisk trunks and uses the asterisk apis to accomplish lots of things ... so you can just throw out asterisk and use a2b..

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